📝 Blog Summary
This guide breaks down why a working TLS handshake tells you nothing about whether your audio is encrypted. It covers what SIP over TLS and SRTP each protect, and when to choose SDES over DTLS-SRTP. You’ll find configuration notes for Asterisk, Kamailio, and FreeSWITCH, plus certificate validation steps teams miss. It closes with troubleshooting for handshake failures and one-way audio.
Ensuring VoIP encryption gets complicated when a call works exactly as expected, but you’re not entirely sure how well it’s protected. TLS is configured, SRTP is enabled, yet one question remains: Is the entire call actually secure, or does the encryption stop short of the audio?
This is where things can get tricky. SIP over TLS protects call signaling, but it doesn’t automatically encrypt the voice traffic. And when SRTP uses SDES, encryption keys can still be exposed if SIP signaling isn’t properly protected.
Getting SIP TLS SRTP configuration right means making sure the right security mechanisms work together across your VoIP infrastructure. Hire VoIP developers with the right expertise to help address these complexities. In this guide, we’ll explore how to configure SIP over TLS and SRTP in production, choose the right key-exchange method, manage certificates, and troubleshoot common issues across Asterisk, Kamailio, and FreeSWITCH.
First, let’s close the gaps between encryption and actual security.
What is the Difference Between SIP TLS SRTP in VoIP?
SIP over TLS and SRTP protect different parts of a VoIP call. TLS encrypts call signaling, while SRTP encrypts the voice and video traveling between endpoints. Using one does not automatically secure the other.
Think of it this way: your signaling can be private while your conversation remains exposed. That’s why VoIP encryption needs both layers.
Why Do You Need Both?
Imagine securing the instructions for a phone call but leaving the conversation itself unprotected. That’s the gap TLS alone can leave.
- TLS without SRTP: Your SIP signaling is encrypted, but the audio may still travel as unencrypted RTP.
- SRTP without secure signaling: Your media is encrypted, but with SDES key exchange, the encryption keys travel in SDP. Without protected signaling, those keys could be exposed.
I’d treat these as two separate checks when reviewing your VoIP encryption setup. A successful TLS handshake doesn’t prove your media is encrypted, and enabling SRTP doesn’t automatically secure your signaling.
The takeaway: SIP TLS SRTP work together to provide comprehensive VoIP encryption. TLS protects call setup, while SRTP protects the conversation, with the right key-exchange method connecting the two securely. For a broader look at these safeguards, a VoIP security enhancements developer’s guide can help you understand how individual security measures fit into a complete implementation.
Next, let’s look at how to choose that key-exchange method.
SDES vs DTLS-SRTP: Which Key Exchange Should You Use?
SDES and DTLS-SRTP both establish SRTP encryption keys, but they do it differently. SDES carries keys in SDP, while DTLS-SRTP negotiates them through a DTLS handshake. Your choice depends on your VoIP platform, security requirements, and whether you need WebRTC support.
SDES for SIP-Based VoIP
SDES (Session Description Protocol Security Descriptions) exchanges SRTP keys through SDP during call setup. It’s supported by many traditional SIP deployments and can be straightforward to configure.
The catch? Your signaling must be protected. If you use SDES without TLS, the keys carried in SDP could be exposed to anyone who intercepts the signaling.
DTLS-SRTP for WebRTC
DTLS-SRTP establishes media encryption keys through a DTLS handshake between endpoints. It’s the standard approach for securing WebRTC media. If your application connects browsers to SIP infrastructure, DTLS-SRTP is generally the right choice for the WebRTC media leg.
Understanding RTPengine for SIP WebRTC media handling can also help clarify how media encryption is negotiated and managed across these connections. You’ll still need to secure SIP signaling separately, often using WebSocket Secure (WSS).
Which One Fits Your Deployment?
One detail worth remembering: encryption doesn’t automatically carry across every leg of a call. Your SBC or media server may need to negotiate separate media sessions and encryption methods.
Before choosing, check what your endpoints and media infrastructure support. A mismatch between encryption methods can prevent calls from establishing or leave media negotiation incomplete.
Once you’ve chosen your key-exchange method, the next step is making sure your server certificates and TLS settings are configured correctly.
Not sure whether SDES or DTLS-SRTP fits your deployment? Our VoIP engineers configure the right key exchange for each call leg.
How to Configure SIP TLS SRTP in Asterisk, Kamailio, and FreeSWITCH
Configuring SIP TLS and SRTP depends on your VoIP platform and how your endpoints negotiate encryption. You need to secure signaling, enable media encryption, and verify that both ends agree on the same settings. Understanding SIP error codes can also help you identify configuration issues during setup and troubleshooting.
The configuration differs across platforms, so let’s look at what matters in each one.
1. Asterisk
In Asterisk, you’ll configure TLS through a PJSIP transport and set media encryption on the relevant endpoint.
Your pjsip.conf needs a TLS transport with a valid certificate and private key. The endpoint configuration then determines how SRTP is negotiated.
[transport-tls] type=transport protocol=tls bind=0.0.0.0:5061 cert_file=/etc/asterisk/keys/fullchain.pem priv_key_file=/etc/asterisk/keys/privkey.pem [secure-endpoint] type=endpoint transport=transport-tls media_encryption=sdes
Choose sdes or dtls based on your endpoint’s capabilities. Confirm the supported options for your Asterisk version before deploying.
2. Kamailio
Kamailio can secure SIP signaling through its TLS module. However, TLS alone doesn’t encrypt the media flowing between endpoints.
For SRTP handling, you can integrate RTPEngine to manage media relaying and encryption negotiation. This is especially relevant when configuring a VoIP network with Kamailio as an inbound and outbound proxy or SBC, where signaling and media encryption must work across different call legs.
Your setup needs to account for:
- TLS listener and certificate configuration.
- RTPEngine media-encryption settings.
- SDP offer and answer handling across call legs.
If your SBC bridges SIP and WebRTC clients, check that each media leg negotiates the correct encryption method.
3. FreeSWITCH
In FreeSWITCH, TLS is configured through a Sofia SIP profile, while SRTP behavior depends on the profile, endpoint, and negotiation settings.
You’ll typically review settings such as tls, certificate paths, and rtp-secure-media in the relevant profile or configuration.
Don’t assume that enabling TLS on a Sofia profile automatically secures media. Verify SRTP negotiation on an actual call, especially when endpoints use different encryption methods.
What Should You Verify After Configuration?
Regardless of your platform, test the complete call path to ensure your SIP TLS SRTP configuration delivers reliable VoIP encryption before going live.
- Confirm that SIP signaling uses TLS.
- Verify that SDP negotiates the expected SRTP method.
- Check for failed negotiations or unencrypted fallback.
- Test calls across different endpoints and network paths.
A call that connects successfully is only the first milestone. Your real test is whether signaling and media remain protected throughout the call.
Your team runs the platform, but nobody owns TLS and SRTP. That’s the gap our VoIP engineers step into.
How to Configure SIP TLS Certificates and Certificate Validation
Your SIP server needs more than a certificate to establish a secure TLS connection. You must also validate the certificate chain, match the server identity, and configure trust correctly. Otherwise, your calls may fail before encryption even begins.
What Should You Configure?
- Obtain a valid certificate: Use a trusted certificate authority (CA). Let’s Encrypt may work for publicly reachable SIP services, depending on your validation method and deployment.
- Configure the certificate chain: Ensure your server presents the required intermediate certificates.
- Match the server identity: Check that the certificate’s Subject Alternative Name (SAN) matches the hostname your SIP clients use.
- Protect your private key: Restrict file permissions so unauthorized users and processes cannot access it.
- Configure CA trust: Ensure your clients trust the issuing CA. Enable client-certificate verification when your deployment requires mutual TLS.
What About SNI and Certificate Errors?
If your infrastructure hosts multiple TLS services on one address, Server Name Indication (SNI) can help select the appropriate certificate. Support depends on your SIP stack and configuration.
A certificate can be valid but still fail verification if the hostname doesn’t match, the chain is incomplete, or the client doesn’t trust the issuer.
Don’t stop at installing the certificate. Test the TLS handshake from the same network and client environment your calls will use.
Once certificate validation works, move on to the next challenge: why secure calls still fail in production.
How to Troubleshoot SIP TLS Handshake Failures and One-Way SRTP Audio
SIP TLS handshake failures usually point to certificate, trust, or protocol mismatches. One-way SRTP audio often involves media negotiation, firewall rules, or RTP routing. The key is to identify where the failure occurs before changing your configuration.
Common SIP TLS SRTP Problems
Where Should You Start?
- Check the TLS handshake. Inspect your SIP server logs and certificate validation results. Confirm that the client trusts the certificate and that both sides support compatible TLS settings.
- Inspect SDP negotiation. If signaling succeeds but audio fails, check the SDP offer and answer. Look for missing crypto attributes, incompatible media profiles, or a mismatch between SDES and DTLS-SRTP.
- Verify the media path. Confirm that your firewall permits the negotiated media traffic. Check RTP routing and any SBC or media relay handling the call. Understanding SBC architecture can help you identify how the session border controller affects media routing and encryption negotiation.
- Check for encryption fallback. Some configurations may permit unencrypted media when secure negotiation fails. Review your platform’s settings and disable insecure fallback where supported.
One detail worth remembering: a successful SIP handshake doesn’t guarantee a successful media session. I would verify signaling, negotiation, and actual media traffic separately before treating the issue as resolved.
Once these checks pass, the next step is to make sure your encryption settings remain secure under real production conditions.
Hardening SIP TLS and SRTP for Production
A configuration that works under normal conditions may behave differently during failover, certificate renewal, or a change in call routing. Production hardening means keeping your encryption requirements intact even when your infrastructure changes.
Here are three areas worth testing before deployment:
- Failover and alternate routes: Check whether backup servers and alternate SIP trunks preserve your TLS and SRTP policies.
- Certificate renewal: Make sure certificate updates don’t interrupt secure connections or leave your server presenting an expired certificate.
- Encryption policy enforcement: Confirm that failed SRTP negotiation cannot silently downgrade calls to unencrypted RTP where your security policy prohibits it.
Your monitoring should also flag repeated TLS handshake failures, unexpected media-encryption changes, and negotiation errors. These signals can help you spot problems before they affect more calls.
The real test isn’t whether encryption works once. It’s whether your system continues to enforce it when conditions change.
Now, let’s bring it all together and secure every call leg.
We secure SIP signaling and media end to end.
The Bottom Line?
Securing VoIP in production takes more than enabling TLS and SRTP. Effective VoIP encryption requires certificate validation, key exchange, media routing, and failover policies to work together to protect every call leg. A single misconfiguration can leave signaling secure while voice traffic remains exposed.
This is why Hire VoIP Developer approaches VoIP encryption with SIP TLS SRTP as an end-to-end security challenge, accounting for the interactions between SIP signaling, media encryption, and the underlying infrastructure.
